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SETU VFXTH
“VoIP – FXO – FXS Gateway”
 Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
 Overview

A Versatile VoIP-FXO-FXS Gateway

A Gateway that provides voice service over IP network using SIP protocol

An effective and flexible solution for accessing Internet based telephone services &

corporate Internet systems across established LAN

Developed to fulfill requirements of SOHO (Small Office Home Office) users & small/

medium scale enterprises

 Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
Interfaces
 Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
 SETU VFXTH Configurations
Configurations VoIP FXO FXS FXO Ports FXS Ports
Channels Ports Ports Label Label

SETU VFXTH0016 16 0 16 0 P01-P16

SETU VFXTH0024 24 0 24 0 P01-P24

SETU VFXTH0032 32 0 32 0 P01-P32

SETU VFXTH3200 32 32 0 P01-P32 0

SETU VFXTH0808 16 8 8 P01-P08 P09-P16

Total 32 SIP Trunks supported in all Configurations

 Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
 SETU VFXTH1616 Hardware Architecture
FXS Modules: Each
Module supports 2
extensions to be FXO Modules VoIP module : CODEC IC &
connected (Total 8) SDRAM. Total 4 such Modules
128
(Total 8) Each supporting 8 channels
MB
RAM

32 BIT RISC
Power PROCESSOR
Supply
OP V : +5V FLASH 32MB
+3.3V,
-27V
-87V CPLD

Input
Supply : DC
Power Jack
24 V, 2.5A

Ethernet Port
 Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
 LED Indications
Total 34 LEDs in SETU VFXTH1616

Power LED : At Power On Power LED will Turn On

(Continuous Green)

32 Port LEDs : FXO and FXS Port LEDS

At Initialization:
P01-P32 : OFF
After approx 16 sec P01-P03 Glow Continuous Red
After approx 20 sec remaining P04-P32 Glow Red Continuous
After 5 Sec : P01 - P32 LED will be Off
 LED Indications
32 FXO/FXS Ports LED indications during normal functioning

Continuously Off Port Idle / Disable

400 ms Red on - Incoming Ring Event

200 ms off -
400 ms RED on -
3000 ms off (2 Blinks)

400 ms on- Off-Hook Event (Dialing State)

400 ms off (continuous) Red

Continuous On Red Speech

 System
LED (STS)
 Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installations Do’s and Don’ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
 Installation DO’s
Dust Proof, Moisture Free Location

Away from electromagnetic Sources

Ventilated Location

Path to Static Charges

Stable Mains Supply

Proper Mains Earth

Proper Telecom Earth

Installation DONT’s
 Overview
Interfaces
Port Configuration
Hardware Architecture
LED Indications
Installation Do’s and Don'ts
Applications
Programming using Phone
Programming using PC
Incoming call Management
Outgoing Call Management
Advance Settings
Maintenance
Status
 Stand Alone Application
2001

FXS1…FXS16

2016
FXO1…FXO16

Ethernet PSTN
Stand Alone Call Possibilities Network

1. FXS  IP Network
2. FXS  PSTN Network IP
Network
Broadband
Modem/Router
 SETU
PBX VFXT1616
In Front of PBX
FXS1…FXS16 Application

FXO1…FXO16
FXS1…FXSN

Ethernet PSTN
FXS ports of SETU VFXTH are Network
connected to FXO Ports of
PBX. Extensions of PBX thus
can avail the PSTN & VoIP
Networks of SETU VFXTH IP
Network
Broadband
Modem/Router
 Behind the PBX Application
SETU
VFXT1616
2001

FXS1…FXS16 PBX

2016 PSTN
FXO1…FXO16
N/W

FXO ports of SETU VFXTH Ethernet FXS1…FXSN

are connected to FXS
Ports of PBX. Extensions
of SETU VFXTH
thus can use the
Trunk of SETU VFXTH IP
Network
Broadband
Modem/Router
Analog Extension PBX Over IP Application
SETU
SETU VFXT1616
VFXT1616
2001 2001
FXO1…
FXS1… FXO16 2016
FXS16 Broadband
Modem/Router
FXO1… PSTN
FXO1… FXO16 N/W
FXO16
Ethernet

Ethernet PBX FXS1…FXSN

IP
Network Broadband
Modem/Router
 Peer to Peer Calling
SETU IP SETU
VFXT1616 VFXT1616
2001 Network
2001
FXS1… FXS1…
FXS16 FXS16
2016

FXO1… FXO1…
FXO16 FXO16

Ethernet Ethernet
PSTN Call over IP (Long Distance converted to Local Call)
SETU SETU
VFXT1616 IP VFXT1616
2001
Network
2001
FXS1… FXS1…
FXS16 FXS16
2016

FXO1… FXO1…
FXO16 FXO16

Ethernet Ethernet

Mumbai Delhi
Programming Using Phone
 Programming Using Phone

Certain parameters of SETU VFXTH can be configured by dialing system commands

from a telephone connected to the FXS port

You can configure certain network parameters like IP address, Subnet Mask,

Connection Type, set the system to default and also view current IP address, Subnet

Mask, Connection Type, DNS and Gateway address by dialing system commands
 SE Login

Connect Analog Phone to FXS port of SETU VFXTH

OFF - Hook the phone

Hear Dial Tone [Toooooooooooooooooooooo]

Dial Command “#19 – SE Password” for Login

Default SE Password is “1234”

Enter System Commands to perform different functions

Dial “00#*” to Exit from Programming mode

 Commands

11 – IP Address – #* (To change IP Address)

12 – Subnet Mask – #* (To change Subnet Mask)

10 – Code – #* (to change the connection type) [1 – static, 2 – DHCP, 3 – PPPoE]

31 – Code – #* (To Enable1/Disable0 VLAN Tag)

51 – Reverse SE Password – #* (To Restore Factory defaults)

 Commands

21 – #* (To view IP Address) & go On – Hook

22 – #* (To view Subnet Mask) & go On – Hook

23 – #* (To view Gateway Address) & go On – Hook

24 – #* (To view DNS Address) & go On – Hook

20 – #* (to view the connection type) & go On – hook

27 – SIP Trunk Number (1 – 9) – #* & go On – hook (To view the status of SIP Trunk)
programming Using pc
 Web Jeeves Login from Local Network

Network Switch

192.168.50.200
SETU VFXTH is located
on Local IP

192.168.50.33
 Web Jeeves Login from Public Network

Internet 203.88.123.231

SETU VFXTH is located

on Global IP

PC with internet
connection
 Web Jeeves Login from Public Network
LAN:
192.168.1.1

Internet

WAN:
203.88.123.231:80 IP : 192.168.1.151
Subnet : 255.255.255.0
Gateway : 192.168.1.1

Router’s port:80 is
forwarded to IP Address of
SETU VFXTH
PC with internet
connection
 Programming

Built – in Web server

GUI based software called Jeeves

Accessible using any web browser

Default IP of Ethernet Port is 192.168.001.136

Default SE password is 1234

Programming
Programming

Enter Ethernet Port IP

Address of SETU VFXTH
Login Page

Enter Password for

Login (Default: 1234)
Home Page
Basic Settings
Network Port Parameters

Select Region of
system installation
and accordingly
Call progress tone
country wise
 Network Port Parameters
• This parameters can be programmed as per existing data network

• Connection type :

1. Static: IP address, Subnet mask & Gateway Address assigned Manually

2. DHCP: IP address, Subnet mask & Gateway Address assigned automatically by

DHCP server

3. PPPoE: Select this option if your ISP provides internet services using PPPoE, If

you select this option you must enter the User ID, password and service name in

PPPoE parameters
 Network Port Parameters

Select connection
type of SETU VFXTH
and according to the
connection type
program the IP details
 Login Password

Password for
Jeeves/FTP/Telnet can be
minimum of 4 characters
and Maximum of 16
characters long
All ASCII characters are allowed except
white space & ( ) ; “ ‘ < > | \ dot (.)
Date – Time Settings

Click on arrow to Set

date and time manually

Set SNTP server address

here to sync date &
time with SNTP server
MWI (Message Wait Indication on SIP Trunk

If you have subscribe for

MWI on SIP trunk for
the voice mail service by
your ITSP then Program
Message retrieval
number provided by
ITSP and port number
ion which MWI is to be
sent
MWI (Message Wait Indication on SIP Trunk
Incoming call management
- SIP trunk
- FXO Port
 Incoming Call Route

• The process of routing calls originated on FXO port and SIP trunks to the

destination port in SETU VFXTH takes place in two steps:

1. Determination of destination number

2. Determination of destination port

Destination Number determination

Outgoing on
which Number
FXO Port
Destination Number Determination on FXO Port

Incoming Call Route

options on FXO Port
Without Any Destination Number

Define destination
port for routing calls

2 different
routings defined
here
1. Route all IC
calls (with
CLI)
2. Route all IC
calls (without
CLI)
Destination Number Determination on FXO Port

• Without any Destination Number

• To the Fixed Destination Number

• On the basis of Calling Party Number

• After answering the call and collecting the digits

 Without Any Destination Number

• Incoming call on the FXO port

• All calls received on the FXO port are directly routed to the fixed destination

port, configured for this port, regardless of the destination number

 Without Any Destination Number

FXO
FXS

022 2631725
2001

SETU VFXTH
No destination number will be provided, only Destination port will be applied
Route To a Fixed Destination Number

• Incoming call on the FXO port

• Call is routed to the Fixed destination number programmed on that particular

trunk line using the Destination port programmed for that trunk

• Destination port can be FXS port, FXO port or SIP Trunk

 Route To a Fixed Destination Number

FXO
SIP

471@matrix- 0265 2630555

pbx.dynalias.org

SETU VFXTH
Fixed Destination Number: 471
 Route on the basis of Calling Party Number

• Incoming call on the FXO port

• Calls are routed to a specific number according to the calling party number

• When there is an incoming call on the FXO port, SETU VFXTH will match the

calling party number with the entries of the calling party number based table,

if a match is found, the call is routed to the destination number

 Route on the basis of Calling Party Number

FXO
SIP

471@matrix- 0265 2630555

pbx.dynalias.org

SETU VFXTH
Calling Number Destination Number
02652630555 471
02226471110 472
After Answering the call & collecting the digits

• Incoming call on the FXO port

• Incoming calls are answered and dial tone is played to the caller, allowing the

caller to dial the desired number

• The number dialed by the caller is considered as the destination number and

dial it out using the destination port programmed

 After Answering the call & collecting the digits
471
Dial Tone

FXO
SIP

0265 2630555

471@matrix-
pbx.dynalias.org

SETU VFXTH
SIP Trunk
Destination Number Determination on SIP Trunk

Incoming Call Route

options on SIP trunk
Destination Port Determination on SIP Trunk

Destination Port
Options on SIP trunk

2 different
routings defined
here
1. Route all IC
calls (with
CLI)
2. Route all IC
calls (without
CLI)
 Destination Number Determination on SIP Trunk

• Without any Destination Number

• To a Fixed Destination Number

• On the basis of Calling Party Number

• To the Called Party Number

 Without Any Destination Number

• Incoming call on the SIP Trunk

• All calls received on the SIP Trunk are directly routed to the fixed destination

port, configured for this port, regardless of the destination number

 Without Any Destination Number

SIP
FXS

022 2631725
2001

SETU VFXTH
No destination number will be provided, only Destination port will be applied
 Route to a Fixed Destination Number

• Incoming call on the SIP Trunk

• Calls are routed to the Fixed destination number programmed on that SIP trunk

using the Destination port programmed for that SIP trunk

• Destination port can be FXS port, FXO port or SIP Trunk

 Route To a Fixed Destination Number

SIP
FXO

0265 2630555 471@matrix-

pbx.dynalias.org

SETU VFXTH
Fixed Destination Number: 0265 2630555
 Route on the basis of Calling Party Number

• Incoming call on the SIP Trunk

• Calls are routed to a specific number according to the calling party number

• When there is an incoming call on the SIP trunk, SETU VFXTH will match the

calling party number with the entries of the calling party number based table,

if a match is found, the call is routed to the destination number

 Route on the basis of Calling Party Number

SIP
FXO

0265 2630555 471@matrix-

pbx.dynalias.org

SETU VFXTH
Calling Number Destination Number
471 02652630555
472 02226471110
 To the Called Party Number

• Incoming call on the SIP Trunk

• Incoming calls are routed to a desired number depending upon the called

number received in the SIP ID of request URI of the INVITE message

 To the Called Party Number

0265 2630555

SIP
FXO

02652630555@
203.88.142.221
0265 2630555

SETU VFXTH
203.88.142.221
Destination Port Determination

Outgoing By
which Trunk
 Destination Port Determination

• SETU VFXTH supports different methods of determining the destination port

for the calls originated on FXS Port, FXO Port and SIP trunks, they are:

1. Fixed

2. On the basis of destination number

3. On the basis of calling party number (Not Supported on FXS Port)

Destination Port Determination on FXS Port

Destination Port
Options on FXS Port
Destination Port Determination on FXO Port

Destination Port
Options on FXO Port
Destination Port Determination on SIP Trunk

Destination Port
Options on SIP trunk
Outgoing Call Management
- FXS Port
- FXO Port
- SIP trunk
FXS Port

For OG Call we can allow

or block outgoing calls,
enable flag to Block the
Outgoing from this trunk
 FXO Port

Enable flag to Block the

Outgoing call from this
trunk, Apply ANT with
Dialed & substitute
number string
 SIP Trunk

Options Related to
Outgoing calls
through SIP trunk
STUN
 STUN (Simple Traversal of UDPs
through NATs)

When the VoIP port (WAN) is located behind a NAT Router & SIP Messages need to

forwarded to the Public Internet

STUN specifies the mechanism required for NAT traversal in SIP messages. STUN

server facilitates traversing through most NATs except symmetric NATs

Illustration of STUN
 Illustration of STUN

STUN Request
STUN Request
Source:192.168.50.161:5060
Source: 115.118.161.163:5060

STUN Response
STUN Response
To:115.118.161.163:5060
Payload:115.118.161.163:5060 To: 192.168.50.161:5060
Payload:115.118.161.163:5060

STUN Server
STUN

Program the STUN Server

IP Address & Port here
STUN

Select NAT type as STUN if you want

to use IP address fetched using STUN
 STUN

Status page will

display the IP
address, port
number and NAT
type fetched
using STUN
Router Public IP Address
 VoIP Port Parameters:
Router’s Public IP Address

Port Forwarding:

Since STUN doesn’t work with symmetric NAT , as an alternative to STUN Port

Forwarding can be done in the router and Router’s Public address that is configured

can be used as Source Port IP Address

Router Public IP Address

Use NAT type as Router

Public IP address
Router Public IP Address

Program Router
Public IP Address here
Router Public IP Address

Status page will display the

Router Public IP address
programmed in the system
parameter page
P2P Call One Device is on Public IP and
Other Device installed behind NAT
Router separates
Port Forward in Private and Public
Router Network

203.88.142.218
Internet Public IP
LAN port of Router WAN
192.168.200.210 203.88.142.221

SETU VFXTH
IP: 192.168.200.195
G/W : 192.168.200.210

Private IP
Router Configuration : Example

Router’s
Network
Parameters

*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.

Router Configuration : Example

Port
Forwarding:
Router’s SIP and
RTP Ports are
forwarded to
Private IP of
SETU VFXTH

*Linksys is a wholly owned subsidiary of Cisco Systems, Inc.

Peer to Peer Calling
 Peer to Peer Calling

• Making an IP call without the intervention of a proxy server is called peer to

peer calling

• As peer to peer calling does not require a proxy server, voice communication

using this application can be done virtually free of cost

• The major cost savings offered by this application makes it a very attractive

mode of inter – branch or intra – office voice communication

Peer to Peer Calling

Enable Program SIP trunk

SIP trunk mode as peer to peer
for peer to peer calling
Peer to Peer Calling

Program the peer to peer table

Click here to delete with destination number &
entry from the table destination address (IP address
of opposite location)
Click here to add new
entry to the table
Proxy Calling
 Requirement for Proxy Calling
Proxy server authenticates the clients for outgoing calls through it

SIP ID
What is required
for
authentication? Authentication ID

Authentication
Password

Registrar Server
Address

Registrar Server
port
Proxy Calling

Enable
the flag
Select SIP Trunk as
Proxy and assign the
authentication
credentials provided
by service provider
 Proxy Calling

If this flag is enabled,

SETU VFXTH will send
the REGISTRAR
MESSAGE to
Registrar Proxy as
applicable
 SIP Registration
On enabling the flag of SIP Registration, following parameters are to be taken care of

This is the time period after which

system will send registration request
to maintain binding with Registrar
Server. Valid range: 00001-65535.
Default:3600 Seconds

When a registration attempt fails,

system resends request to registrar
server after this timer’s expiry. Valid
range: 00001-65535. Default:10
Seconds
SIP Registration

System will get unregistered

with the current server & will
register with the alternate
server, if fallback occurs while
sending INVITE message when
Switch Registration to
Alternate Server on Fallback is
enabled
 Registrar Settings

If you want the

system to send
DNS SRV query to
the configured
domain server,
enable this flag
 What is DNS SRV?

Dialing by domain names lets a SIP user have a single public “SIP Address” which

can be redirected at will to their current location.

SRV records maintain stability and also opens up the possibility to use your own

domain, regardless of the domain of the SIP service you are using
 SIP Registration

Enable the flag, if

your service
provider supports
multiple servers
in its network
Advance Settings
 Access Codes

• Access code is a string of digits dialed to use a feature

• SETU VFXTH users can access the features and facilities by dialing the access
code assigned to them from a phone. User can

1. Enable/Disable a feature

2. Access Supplementary feature

3. Enter into the programming mode

• SETU VFXTH provides default access code for all features, you can change it to
suit your preferences
 Access Codes

Access codes can be

changed from here
 Access Codes

Access codes can be

changed from here
 Allowed – Denied Numbers

• This feature provides the flexibility to allow or deny dialing of a particular

number or a set of numbers from a particular port or all ports

• Allowed Denied number logic makes use of two number lists:

1. Allowed Numbers List: this is the list of numbers that can be dialed out from

the SIP trunk (default number list – 7)

2. Denied Numbers List: this is the list of numbers that are to be restricted from

being dialed out from the SIP trunk (default number list – 8)
Allowed – Denied Logic on FXS Port

Apply allowed denied list on FXS

Port & program the number list
for allowed & denied numbers
Allowed – Denied Logic on FXO Port

Apply allowed denied list on FXO

port & program the number list
for allowed & denied numbers
Allowed – Denied Logic on SIP Trunk

Apply allowed denied list on SIP

trunk & program the number list
for allowed & denied numbers
 Automatic Number Translation
• This feature is used to translate the dialed number string to preprogrammed

number string

• ANT can be used to modify, add or delete the prefix of the destination number

string

• For this feature we need to configure dialed number string and substitute

number string in number list table

• ANT feature is applied on destination ports (On all SIP trunks and FXO Ports)
Automatic Number Translation

Apply ANT on FXO port and

program the table number
Automatic Number Translation

Apply ANT on
SIP Trunk and
program the
table number
Automatic Number Translation

ANT table

Examples on
how to program
 Black Listed Callers

• SETU VFXTH supports feature ‘Black listed Callers’ which enables you to block

incoming calls from specific numbers and addresses on the SIP trunks

• This feature is applicable on source port only

• To use this feature, user must configure the numbers of unwanted callers in a

number list

• Enable the Reject Calls from Blacklisted Caller check box on the SIP trunks on

which you want to apply this feature

 Black Listed Callers

Apply black listed

caller feature on
selected SIP trunk and
define the number list
for the same Black
Listed Callers
 Call Detail Record (CDR)
• It’s a record for the calls, containing information about the gateway’s usage

when call was made

• Maximum of 2000 call record entries can be stored

• Call record entries are stored in FIFO logic

• User can set different filters as required and generate Call Detail Record (CDR)

report

• Call records can be cleared manually at any time

 Call Detail Record (CDR)
• It is possible to get following details of a call with CDR
1. Date of call origination
2. Time of call origination
3. Calling number
4. Called number
5. Duration of call
6. Source port
7. Destination port
8. Disconnected by
9. Cause
10. PIN number
11. Remarks
 Call Detail Record (CDR)
• Below mentioned filter can be programmed for CDR
1. The port from which the calls originate (Source Port)
2. The port on which the calls terminate (Destination Port)
3. Calls made on particular dates
4. Calls made at a particular time
5. Calls of a certain duration
6. Calls of certain called party numbers
7. Calls of certain calling party numbers
8. Calls made with PIN authentication
9. Calls made without PIN authentication
Call Detail Record (CDR)
Set filter parameters
for CDR here

Click on download to get Zip file

Click here to clear containing CDR in .csv and .txt format
all call records
Call Detail Record (CDR)

CDR can also be

viewed from Jeeves
 PIN Authentication
• PIN authentication is a security feature to restrict access to the system and

prevent possible misuse of resources

• User can use the PIN authentication on the source port to establish identity of

callers before their call is processed by SETU VFXTH

• PIN authentication can be used on the source port only if the incoming call

routing for the source port is set to After answering the call and collecting digits

• To use this feature it must be enabled on the source port and the PIN

authentication table must be configured

 PIN Authentication
• The PIN authentication table stores up to 500 PIN numbers and their

corresponding authentication passwords

• If PIN authentication is enabled on source port, SETU VFXTH answers the

Incoming call and plays a feature tone, it waits for the caller to dial the PIN

number and password, it matches them with the PIN authentication table, if

match is found it allows the call to be processed

• In case of wrong PIN entered, SETU VFXTH allows the caller to make two more

attempts, if the caller fails to dial correct PIN and password in all attempts, the

system disconnects the call

PIN Authentication – FXO Port

Select routing type

‘after answering the
call and collecting
the digits’ for PIN
authentication
feature to use

Enable this flag for

prompting caller
to enter PIN
PIN Authentication

Enter PIN number & PIN password,

system checks PIN entered by the caller
during call with the entries in the PIN
authentication table, if match found then
only the call will be processed further
 Digest Authentication

• Digest authentication is a challenge – based authentication service of SIP to

authenticate the identity of the originator of SIP request in the INVITE message

• The recipient of the request can ascertain whether or not the originator of the
request is authorized to make the request

• When the digest credentials of the originator – User Name and Password – in
the INVITE message are authenticated and accepted by the recipient, the
originator and recipient are connected

• You may use the digest authentication to restrict access to SETU VFXTH to
specific callers, prevent unwanted or malicious calls
 Digest Authentication

• When this feature is enabled on a SIP trunk for all Incoming calls

1. SETU VFXTH will challenge the identity of the calling party

2. When the calling party sends its credentials, SETU VFXTH authenticates the

credentials by matching it with its Digest Authentication table

3. If a match is found, the calling party will be authenticated and the call will be

allowed on the SIP trunk

4. If no match is found, SETU VFXTH will consider it as invalid authentication

information and reject the call

 Digest Authentication

Enable apply flag in

SIP trunk to use
digest authentication
Digest Authentication

Enter Digest credentials

(User ID and User
Password) of calling party
 Static Routing

• Static Routing Table is required when you have more than one router (Gateway)

in your network and you want SETU VFXTH to send packets to multiple

routers/gateways for different types of calls

• If you have only one router connected in the network , you need not configure

this table & LAN interface of router will act as the default gateway for the system
Static Routing

Program the static routing table with

the details, if the match is found here
then gateway will send the packets to
defined gateway address opposite to
the destination address
Prefix to Domain Name Conversion
• Prefix to domain name conversion is used when a user sets call forward or

makes a blind transfer on SIP, this feature is applicable only when the

destination port is SIP

• SETU VFXTH supports multiple SIP trunks & FXS ports, when a FXS port user dials

a SIP number, SETU VFXTH routes the call to the IP network using the SIP trunk

determined by the routing mechanism. The FXS user can dial only numbers not

domain names, therefore it becomes necessary that the domain names be

assigned prefix codes which the FXS user can dial

Prefix to Domain Name Conversion

• User need to program prefix v/s domain name in the table

• This table is not checked for making an outgoing call, but it is checked when

some FXS port has set call forward and only number is programmed or user is

doing blind transfer

• For example prefix in the table is programmed as *123 and domain name as

abc.com and destination number for call forward is *1239974 then it will be

replaced by 9974@abc.com
Prefix to Domain Name Conversion

Define prefix and

domain name in
the table
 Disconnect Tone

• If call disconnection is signaled by your CO network in the form of disconnect

tone on the FXO Ports

• You must enable Disconnect Tone Detection on the FXO port and select the

Disconnect tone type

• To enable the system to detect the disconnect tone accurately, you must

configure the cadence and frequency of the disconnect tone type you selected,

as supported by the CO network

 Disconnect Tone

Enable disconnection
tone detection here
Disconnect Tone

Program the disconnect

tone cadence here
 Emergency Numbers

• SETU VFXTH supports dialing of emergency numbers from all ports, Emergency

numbers and their respective routing groups must be configured in the

emergency number table

• User can configure up to 10 numbers of emergency services such as ambulance,

fire brigade, police etc.

• By default, No emergency numbers are loaded in the system, in the emergency

number table
Emergency Numbers

Click here to Edit

entry of the table

Click here to add new

entry to the table Click here to delete
entry from the table
Features
 Class Of Service

• If any FXS port want to use supplementary services then these services must
be activated in COS for particular FXS port as well as at SIP services provider in
case of SIP account calling

• SETU VFXTH offers following telephony features, which they can access by
dialing access codes
1. Call Hold 6. Blind Transfer
2. Call Forward 7. Attended Transfer
3. Call toggle 8. Do Not Disturb (DND)
4. Call waiting 9. Hotline
5. Conference
 Class Of Service

Enable required
feature from Class
of service on
particular FXS port
Supplementary Services

Enable the
supplementary
services after
enabling the
feature in COS
 Subscriber Type

• When SETU VFXTH is interfaced with service provider server – ITSP or other

PBX that supports supplementary services that require dialing of Flash like call

hold, call transfer, call waiting, you must select the subscriber type according to

the extent of feature access you want on the FXS port connected to the system
 Subscriber Type

• Select Network if you want to use supplementary services supported by the

other PBX, you can access the service provider features by dialing FLASH, you

will not be able to access the local features of SETU VFXTH

• Select Gateway if you want to use supplementary services supported by the

SETU VFXTH, in the gateway mode you will also be able to access the

supplementary services of the service provider which require dialing of FLASH

Subscriber Type

Select the
subscriber type
of your choice
 FXS Port
Signaling Loop Start

Connector RJ-45

Off-Hook Line Impedance 600 Ω / 900 Ω / Complex

No. of Long Loop Extension 4

Loop Limit 1800 (Max) Excluding Telephone Set

On-Hook Voltage (Tip/Ring) -48 V

Off-Hook Current 25 mA (Max)

Ringing Voltage Trapezoidal 60 VRMS/25Hz and

Sinusoidal 52VRMS/25Hz
 FXS Port
REN 3

DTMF Detection ITU-T Q.24

CLI Presentation DTMF, FSK ITU-V23 & FSK Bellcore

Protection Over Voltage Secondary Protection

Return Loss >18 dB

Longitudinal Balance >50 dB

Transmission Level Adjust Tx Gain : -3dB to +6dB; Rx Gain : -3dB to +6dB

Answer Signaling on FXS Battery Reversal

Disconnect Signaling on FXS Battery Reversal & Open Loop Disconnect

 FXS Port

Hardware settings
on FXS port
 FXS Port

General settings
on FXS port
 FXS Port

First Digit & Inter

Digit wait timer
 FXS

 First Digit Wait Timer:

• Signifies the time for which the system waits for receiving a first digit after

going off – hook from FXS port

• On expiry of this timer, system will give error tone to the user

• It is programmable from 01 to 99 seconds (Default: 15 seconds)

 FXS

 Inter Digit Wait Timer:

• Signifies the time period between 2 consecutive digits while the system is

receiving the digits from caller

• On expiry of this timer, ATA1S will process the digits dialed so far by the user

• it is programmable from 01 to 99 seconds (Default: 5 seconds)

 FXO Port

Return Loss >18 dB

Longitudinal Balance >50 dB

Transmission Level Adjust Tx Gain: -15 dB to +10 dB

Rx Gain: -15 dB to +10 dB

Call Maturity Delay & Polarity Reversal

Answer Supervision on FXO Battery Reversal

Disconnect Supervision on FXO Battery Reversal & Open Loop Disconnect

 FXO Port
REN 3

DTMF Detection ITU-T Q.24

CLI Presentation DTMF, FSK ITU-V23 & FSK Bellcore

Protection Over Voltage Secondary Protection

Return Loss >18 dB

Longitudinal Balance >50 dB

Transmission Level Adjust Tx Gain : -3dB to +6dB; Rx Gain : -3dB to +6dB

Answer Signaling on FXS Battery Reversal

Disconnect Signaling on FXS Battery Reversal & Open Loop Disconnect

 FXO Port

Hardware settings
on FXO port
 FXO Port

General settings
on FXO port
Making a new call using access code

• This feature enables callers to disconnect the current call and make a new call
using SETU VFXTH without getting disconnected from the system

• This feature is useful when you want to make multiple calls without getting
disconnected each time their call ends

• This feature is applicable only on the FXO port and only when After answering
the call and collecting digits is selected as the destination number
determination method

• If you have enabled Connect source port when number is out dialed on the FXO
port, you will not be able to provide this feature to callers
Making a new call using access code

• To make a new call using access code

 In speech with the current call

 Dial #91

 Current call will disconnect

 Dial the new number you want to call

 Speech will be establish on the new call as called party answers the call

 While in speech dial #91 again to make another new call

Making a new call using access code

Enable the flag to allow

user making new call
using access code
 Disconnecting a call using access code

• SETU VFXTH enables user to disconnect a call using an access code

• When the call disconnect access code is dialed, SETU VFXTH releases the port
engaged in the call

• This feature is applicable only when destination number determination method

is selected as After answering the call and collecting digits

• If you have enabled Connect source port when number is out dialed on the
FXO port or have enabled Connect source port when 183 is received on SIP on
the SIP trunk, you will not be able to provide this feature to users
Disconnecting a call using access code

Enable the flag to allow

call disconnection using
access code
Disconnecting a call using access code

Enable the flag to allow

call disconnection using
access code
 IP Dialing
• SETU VFXTH supports direct dialing of IP addresses from the source port. To
provide IP dialing facility to the users, you must configure a SIP trunk or a SIP
group for IP dialing

• IP number can be dialed with dot ’.’ as entered by ‘*’ while dialing it

• For e.g. to dial IP address 192.167.100.1 dial as 192*167*100*1 from the

Phone at FXS

• When an IP address is dialed from the source port of SETU VFXTH, the system
does not check the destination port determination method you have
configured for that port, instead it routes the dialed IP address through the SIP
trunk or SIP group you configured for IP dialing
IP Dialing

SIP trunk or
SIP trunk
group can be
defined for IP
dialing
 100rel and SIP PRACK

SIP PRACK (SIP Provisional Acknowledgement) is a method to enable reliability

for SIP 1XX messages

Generally PRACK message flows from Calling Party to Called Party

The Called Party answers the PRACK by 200 OK and PRACK is only for 1XX
messages other than 100 Trying
System Parameters

Enable Provisional
acknowledgement
for all 1xx messages
other then 100 trying
 SIP Timers

• SIP Invite Timer

• SIP Provisional Timer

• General Request Timer

 SIP Invite Timer

• It is the time for which SETU VFXTH waits for a response from the called party

after sending INVITE message

• This time starts after sending INVITE message to the called party and stops on

receipt of provisional response or final response or when the user goes ON-

Hook, on expiry of the timer the call is disconnected

• The range of SIP INVITE Timer is 10 - 80 seconds (Default: 30 Seconds)

 SIP Provisional Timer

• It is the time for which SETU VFXTH waits for final response after receiving

provisional response from the called party

• This timer starts on receipt of provisional response from the called party and

stops on receipt of final response from the called party or when the user goes

ON-Hook, on expiry of the timer the call is disconnected

• The range of SIP Provisional Timer is 10 - 180 seconds (Default: 60 Seconds)

 General Request Timer

• It is the time for which SETU VFXTH waits for the response of a transaction

request

• This timer starts on initiating a transaction

• This timer stops on receipt of a response for the request

• On expiry of timer, the SETU VFXTH clears the transaction

• The range of SIP Provisional Timer is 10 - 60 seconds (Default: 20 Seconds)

System Parameters

Program the timer

values according to
the requirement
 SIP Over TCP

• The SIP over TCP option allows you to send/receive the SIP messages over TCP

• SIP over TCP is applicable for both Proxy and Peer to Peer

• By Default SIP messages transported over TCP

• Disable the flag to send SIP messages over UDP

 SIP Over TLS

• The SIP over TCP option allows you to send/receive the SIP messages over TLS.

TLS protects SIP signaling against loss of integrity, confidentiality and against reply

• SIP over TLS is applicable for both Proxy and Peer to Peer

• By Default SIP over TLS is enabled

• Disable the flag to disable SIP over TLS

System Parameters

Program SIP TCP, UDP and TLS

port values. Default: 5060 for
TCP, UDP and 5061 for TLS
Server Port

Server ports can be

changed to any value
from 1021 to 65,535
Management/Security

Server ports can be

changed to any value
from 1021 to 65,535
Certificate
 Certificate

• SETU VFXTH supports certification for TLS, Web Server, Firmware Upgrade,

Configuration Upgrade and TR-069.

• SETU VFXTH supports two types of Certificates: Self-Signed Certificate and CA

Signed Certificate.
 Self – Signed Certificate

• A self-signed certificate is created by the clients themselves or by the Servers and

then given to their clients.

• It means that you yourself become the Certificate Authority (CA), create a CA
Certificate and sign it.

• The self-signed certificate is faster to create but is not signed by a trusted CA

Organization.

• The self-signed certificate must be installed in the trusted list of clients that
connects over TLS with the Server. Because the certificate has been self signed, the
signature is not likely to be in the clients’ trust file, hence, they need to add it.
 Certificate
Generate self signed CA
certificate by entering the
required details below

Click generate
to generate new
certificate for
entered details

Once you generate self-signed certificate, you must send it to your clients so that
they install it in their trusted list.
Certificate

System will show

generated
certificates under
trusted root CA
 System Certificate

• After creating a Self-Signed CA Certificate, you can either,

• Generate a System Certificate for your clients. These System Certificates can
then be given to the respective clients OR
• The Clients can prepare their own System Certificates. For this you need to
send them the CA Certificate created by you OR
• Generate a Certificate Signing Request (CSR), if you want the Certificate to be
signed by a third party
If the clients prepare their own certificates, you need to send your CA Certificate to
all the clients. The clients must upload the same in their system. Similarly, all the
clients must send their CA Certificates to you and you must upload the same in
your system. To avoid this, it is recommended that you create the Certificates and
then provide it to your clients
 If you want to get a CA Signed Certificate, you need to do
the following:
1. Generate and enroll the Certificate Signing Request (CSR).
2. Get the Certificate Signing Request (CSR) verified and
signed by the Certified Authority (CA).

Enter details to
generate system
certificate
Certificate
User can also upload
the certificates

List of available
system
certificates
Certificate

Define the certificate

to be used for
desired application
Maintenance
 Firmware

Program the details

for Auto firmware
upgrade

Upgrade firmware
automatically from
Matrix Server
Browse the ZIP file having
new firmware files & click on
Upgrade button to upgrade
the system firmware
 Configuration

Program the details

for Auto configuration
Browse the ZIP file having
configuration files & click on Click on Backup
Upgrade button to upgrade Configuration to
the system configuration save config.zip file
 System Debug
• Debugs are logs of actions and events that take place on system, these logs are
useful for troubleshooting and system security

• SETU VFXTH supports Syslog client for debugging, Syslog client enables the
system to send debug messages in Syslog format to the remote ‘Syslog server’
on the IP network

• Syslog uses the UDP as transport protocol

• To be able to use this feature, you must enable ‘Syslog’, configure the Syslog
Server Address and define the server port on which the Syslog will listen for
debug messages
 System Debug

ug events
be viewed
he screen
Click debug settings to set
parameters for debug and
to start debug in PC/Laptop
connected to SETU VFXTH
System Debug

Program the IP address and

port number of PC/Laptop
where Syslog server is installed

Debug for Port: clear the check

box to disable the debug for the
port which is not needed
 SNMP

• SNMP – Simple Network Management Protocol

• SNMP protocols supported – SNMPV1, SNMPV2C, SNMPV3

• SETU VFXTH is having built in SNMP Server (SNMP Server). It receives SNMP

requests and generates SNMP responses or notifications

• SNMP Manager usually network management station. It generates SNMP

requests and receives SNMP responses and notifications. The SNMP manager is

an SNMP client
 SNMP

Program
SNMP
details
 System Port Activity

System port activity

like Idle, Inactive,
Disable, Dial, Speech,
ringing, Incoming Call
Proceeding, Remote
Held, Error
 PCAP Trace

• PCAP or Packet capture consists of intercepting and logging the traffic passing

over the network, PCAP intercepts each packet in the data streams that flow

across the network, and can decode and analyze its contents

• A maximum 2MB of packets can be captured and stored in the system

• SETU VFXTH also supports filters and promiscuous mode for capturing packets

• If promiscuous mode is enabled, SETU VFXTH will capture all network traffic and

if disabled then system will capture only traffic that is directly related to SETU

VFXTH (to or from SETU VFXTH)

 PCAP Trace

Enter the filter

Click here to Enable details here
Promiscuous mode

Once the PCAP is captured save

the trace file on your PC/Laptop

Click here to start

the PCAP trace Click here to stop
the PCAP trace
 Manual Call Test

• Select source port and destination port with source number and destination

number.

• When Call button from GUI is pressed system will call source number first and

when answered by source port it will ring on destination port & speech path

can be checked

• Clicking on call button will also lead the programmer to system port activity

page to monitor the status of the port during call progress

Manual Call Test
 AC Impedance Test (FXO)
• SETU VFXTH supports the AC Impedance Test for clear, audible and echo-free
speech over FXO Ports.

• This test helps you to set the most appropriate values for the FXO Port
Parameters —AC Termination Impedance, CO Termination and CO Line Type— to
correct the line impedance mismatch between the AC Termination Impedance
presented by the FXO Port of SETU VFXTH to the line and the CO Termination
Impedance presented by the Central Office to the line.

• While the test is being conducted, you will hear pulsating tone on all the ports of
the system. (Mute the microphone of destination landline number or mobile
number when call is answered by destination number)
AC Impedance Test (FXO) Enter phone number on which
system will make the call in
order to complete the test

For more details

click help
Select the FXO port
on which you want
to run the test

Click on start
test and wait
for the results
Default System

Click OK to factory
default the gateway
Soft Restart

Click OK to Restart
SETU VFXTH
 TR – 069
• TR-069, also known as CPE WAN Management Protocol (CWMP), is a remote
management protocol used for secure communication between the Customer
Premises Equipment (CPE) and an Auto-Configuration Server (ACS) for various
functionalities such as:
 Auto-configuration and dynamic service provisioning
 Firmware Management
 Status and performance monitoring
 Diagnostics

• SETU VFXTH supports TR-069. Using TR-069, service providers can manage SETU
VFXTH remotely for the functions described above.
 TR – 069

Program
TR-069
details
STATUS
System Detail

Version Revision
details
Firmware Status

Last Firmware up gradation

details if scheduled firmware
upgrade is ON
Configuration Status

Last Configuration up
gradation details if scheduled
firmware upgrade is ON
 Network Status

IP details status
of IP configured
in SETU VFXTH
 FXO Port Status

Line connection
status on FXO
 SIP Trunk Status

SIP trunk Status